I see in looking the invite is coming in on port 5060. Normally with
sipxbridge the invite would come in on port 5080, with remote users
and your SRV records coming in on port 5060.

If you are supporting remote users this would cause an issue with media relay.

Can you explain your configuration?

I might suggest you look at this drawing so you understand how someone
here would normally see a call flow...

http://blog.myitdepartment.net/wp-content/uploads/2009/11/Call-Setup-Example-sipXecs-through-ITSP1.pdf

On Sat, Jun 12, 2010 at 6:16 PM, Alexander Shvaryov <[email protected]> wrote:
> Hi All,
>
> I got problem with inbound audio (no audio) when call route in DID.
>
> My settings:
> - SipXecs v.4.2
> - Sip trunk to ITSP (with SipXbridge)
> - "Incoming calls destination" = empty (inbound calls are directly
> routed to the specified number)
> - DID (8127507501) - alias for extension (202)
>
> If call first route in AA (Incoming calls destination = operator) and
> after to extension - inbound audio correctly.
>
> I suppose that cause - no SDP in 200 Ok (attached SIP logs)
> http://www.ietf.org/mail-archive/web/sip/current/msg00857.html
>
> I will be happy if you give me some direction what to look
> troubleshooting this problem.
>
> Thanks in advanced!
>
> PS: Sorry, my English is bad yet.
>
>
> --
> ------------------------------
> Thanks,
> Alexander
>
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>



-- 
======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: [email protected]
Fax: 434.984.8431

Email: [email protected]

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